DSL Integrated Call Waiting

ABSTRACT

One embodiment of a call handling method, among others, includes receiving a query from a switching system indicating that a called party telephone is busy; examining handling information for the called party to determine if the called party has voice over Internet Protocol (VoIP) service; and instructing the switching system to then route the analog call over a VoIP connection to the called party after the called party chooses to communicate using the VoIP service.

CROSS-REFERENCE TO RELATED APPLICATION

This application is a continuation of copending U.S. utility applicationentitled, “DSL Integrated Call Waiting,” having Ser. No. 10/159,306,filed May 30, 2002, which is entirely incorporated herein by reference.

FIELD OF THE INVENTION

The present invention is generally related to telecommunications andmore particularly to the integration of telephone systems and digitalsubscriber lines.

DESCRIPTION OF THE RELATED ART

The telecommunications industry has undergone rapid changes in the pastseveral years. With the development of the advanced intelligent network(AIN), telephone companies are poised to offer a multitude of newservices to subscribers. Typically, AINs provide a more flexiblehandling of telephone calls. This flexibility is provided by a complexpacket-switched network which allows for high speed communication andhigh volume traffic. One example of an AIN is further described in U.S.Pat. No. 5,701,301 and U.S. Pat. No. 5,838,774, each of which is herebyincorporated by reference.

AIN embodiments in the United States use a signaling system 7 (SS7)protocol to transport messages. Instead of circuit switching, the AINrelies on soft switching to provide high speed routing for telephonecalls. The AIN comprises service switching points (SSPs), service nodes(SNs), signal transfer points (STPs), and service control points (SCPs).An SSP is typically an AIN-compatible switching office. The SN is asmart termination device that assesses incoming call information andmake appropriate connections. The SSPs are connected by a number of STPswhich transfer data among the SSPs and between the SSPs and SCPs. TheSTPs can generally be described as the routers which read the packet andtransfer it to the called party SSP. Finally, the SCP is typically afault tolerant computer that is coupled to a central database. Thiscentral database comprises a host of subscriber and routing information.

For better understanding a call routing sequence on an SS7 network willnow be described. Typically, when a call is placed a calling party dialsa telephone number and an SSP receives the place call request and routesit to the proper SSP associated with the called party. When the calledparty SSP receives the call request, it causes a trigger to fire. Thistrigger then causes the SSP to send a query across the STPs to an SCP.The query typically comprises asking the SCP how the call should behandled, such as specific subscriber instructions and any other specificrouting information that is necessary. After receiving handlinginformation from the SCP, the SSP uses these instructions to create apacket to send across the STPs to the called party SSP. The called partySSP then triggers and asks the SCP for subscriber-specific handlinginformation for the called party. Typically the SCP will merely instructthe SSP to connect the call, however, the called party may have specialinstructions for incoming calls. However, these instructions havetypically not included a reasonable mechanism by which to connect anincoming call while retaining a connection to a call already connected.

Therefore, there is a need for systems and method that address theseand/or other perceived shortcomings of the prior art.

SUMMARY OF THE INVENTION

Embodiments, among others, of the present disclosure provides a callhandling system. One embodiment of a call handling system includes ahandling database system which tracks customer specific subscriptioninformation and routing and handling information for telephone serviceprovider customers. The handling database system looks up handlinginformation for a called party and instructs a switching system toconnect an analog call from a calling party to the called party. Thehandling database system receives a query from the switching systemindicating that a called party telephone is busy, and in response, thehandling database system examines the handling information for thecalled party to determine if the called party has voice over InternetProtocol (VoIP) service and instructs the switching system to then routethe analog call over a Internet connection to the called party after thecalled party chooses to communicate using the VoIP service.

Embodiments also include a call handling method. One embodiment of acall handling method, among others, includes receiving a query from aswitching system indicating that a called party telephone is busy;examining handling information for the called party to determine if thecalled party has voice over Internet Protocol (VoIP) service; andinstructing the switching system to then route the analog call over aVoIP connection to the called party after the called party chooses tocommunicate using the VoIP service.

Other systems, methods, features, and advantages of the presentinvention will be or become apparent to one with skill in the art uponexamination of the following drawings and detailed description. It isintended that all such additional systems, methods, features, andadvantages included within this description, be within the scope of thepresent invention, and be protected by the accompanying claims.

BRIEF DESCRIPTION OF THE DRAWINGS

The invention can be better understood with reference to the followingdrawings. The components in the drawings are not necessarily to scale,emphasis instead being placed upon clearly illustrating the principlesof the present invention. Moreover, in the drawings, like referencenumerals designate corresponding parts throughout the several views.

FIG. 1 is a block diagram illustrating a first embodiment, among others,of the present invention.

FIG. 2 is a flowchart showing the operation of the first embodiment ofthe present invention, among others.

FIG. 3 is a schematic diagram illustrating one embodiment, among others,of the present invention.

FIG. 4 is a call flow diagram detailing the operation of the embodimentshown in FIG. 3.

FIG. 5 is a schematic diagram illustrating an alternative embodiment ofthe present invention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

The invention now will be described more fully with reference to theaccompanying drawings. The invention may, however, be embodied in manydifferent forms and should not be construed as limited to theembodiments set forth herein; rather, these embodiments are intended toconvey the scope of the invention to those skilled in the art.Furthermore, all “examples” given herein are intended to benon-limiting.

Referring now to FIG. 1, shown is a block diagram of one embodiment,among others, of the present invention. In this embodiment a callingparty location 100 includes a standard plain-old telephone service(POTS) telephone 101 operating substantially within the POTS frequencyrange. The POTS frequency range is typically defined as the frequencyrange of 0 to 4 kilohertz (kHz), which contains substantially the rangeof frequencies which are audible to the human ear.

The phone 101 is typically connected to a switching system 102 by aconnection 103. The switching system routes calls placed by a callingparty to a called party telephone 104. When a digital subscriber line(DSL) is present, the called party telephone is connected to theswitching system 102 through connections 105, 106, andmultiplexer/demultiplexer 107. The called party telephone is typically astandard POTS telephone and resides at a called party location 108. Theswitching system 102 typically includes a network of switches which areconnected to a call processing system 109 through a connection 110. Themultiplexer/demultiplexer 107 could also be viewed as being containedwithin the switching system 102.

The called party DSL connection can include a computer 111 coupled to amodem 112. The modem 112 typically allows the computer 111 to send andreceive information by coding data in a frequency range that is abovethe POTS frequency range. In this way, both the phone 104 and thecomputer 111 can use a single subscriber line. Themultiplexer/demultiplexer 107 then receives the multiplexed signalthrough connection 106 and demultiplexes the signal, sending the highfrequency data signal to the network 113 across connection 114, and thePOTS frequency signal to the switching system 102 across connection 105.

The computer 111 may also include a conventional internet protocol (IP)phone 115. The IP phone 115 typically sends voice information on thehigher frequency data signal created by the modem 112. Thus, undernormal conditions the IP phone 115 is required to be connected toanother IP phone, since a POTS frequency range device generally cannottranslate the higher frequency voice information into the audible POTSfrequency range.

Alternatively, the IP phone 115 could be a “soft phone.” As known tothose skilled in the art, soft phones are not physical phones, but aremerely a software representation of the functionality of a physicalphone running on a computer, typically utilizing a speaker andmicrophone of the computer. Often the soft phone has a graphical userinterface (GUI) based upon the appearance of a physical phone, whichfacilitates use of the soft phone. However, it is not necessary that aGUI be used to represent the soft phone nor is it necessary that the GUIis based upon the appearance of a physical phone.

During operation, the switching system 102 can receive call routinginformation which is stored on the call processing system 109. Further,the call processing system 109 can store any subscriber-specificinformation related to call handling. In other words, if the calledparty has subscribed to caller-identification (caller-ID), the callprocessing system 109 instructs the switching system 102 to include acaller-ID information packet with the call notification sent to thecaller on line 105. The call notification typically includes an auralprompt such as ringing, however, it can include other sensory prompts,such as vibration, among others. A number of other services can beprovided to subscribers. A few of these include call-return,call-forwarding, call-waiting, call-waiting disabled, and caller-IDblock, wherein a called party can subscribe to a specific service thatcan be provided to the called party without necessarily providing theservice to all users of the phone system.

Another service that can be provided to the called party is a DSLintegrated call waiting service, which is embodied in the presentinvention. Typically, when the calling party places a call, the switchin the switching system 102 that is associated with the calling partycan, as necessary, query the call processing system 109 and can routethe call to the switch associated with the called party. The switchassociated with the called party can receive the call, but a terminationbusy (T_Busy) trigger will fire when the called party telephone 104 isbusy.

The T_Busy trigger causes the switch to send a query to the callprocessing system 109 to request handling information. The callprocessing system 109 then checks a subscriber database and finds thatthe called party subscribes to the DSL integrated call waiting service.When the called party subscribes to the DSL integrated call waitingservice, the call processing system 109 can search the network 113 viaconnection 116 to find out whether the called party has a computer 112with an active DSL session on line 114, throughmultiplexer/demultiplexer 107.

Alternatively, the computer 111 could include a client application that,upon activating a DSL session, registers the computer 111 with the callprocessing system 109. Thus, the call processing system 109 could merelysearch its own registration database to ascertain whether an active DSLsession is available at the called party computer 111.

When there is an active DSL session at the called party computer 111,the call processing system 109 can obtain the internet protocol (IP)address of the called party. After obtaining the IP address, the callprocessing system 109 can push information to the called party computer111 IP address. An application, such as a casting client or a chatclient, on the called party computer 111 could facilitate theinformation being pushed, or the called party computer 111 could be setup to accept the pushed information from the call processing system 109.

The information sent to the computer 111 can be obtained through anumber of different sources, such as a customer database, internetsearch, or even through an agreement with an internet service providerto use their customer database. In one example, the information wouldtypically at least include a selectable prompt that textually askswhether the called party would like to connect the telephone call overthe DSL connection. Alternatively, the prompt could further include orbe accompanied by other selectable options, such as hold, send tovoice-mail, playing a customized busy announcement, sending a shortinstant messaging (IM) message to the caller, disconnect current phonecall and connect incoming call, ignore incoming call, etc. Theinformation could further include personal information of the callingparty. In one embodiment, among others, this personal information couldinclude a name and telephone number of the calling party, such as isdone with standard caller-identification. However, the personalinformation could also include the address of the calling party, or auniversal resource locator (URL) corresponding to a web page of thecalling party, or even a photograph of the calling party.

Further, in one embodiment, among others, the present invention includesa system whereby the calling party can control what information is sentto the calling party. The calling party could access a personalinformation file through the internet and adjust his/her personalprofile to include or exclude address, web pages, telephone number, etc.Thus, when the calling party places a call, the information included inthe personal profile can automatically be sent to the called party.Alternatively, the calling party may be given access to adjust his/herpersonal profile via an interactive voice response menu system, atelephone service provider operator, or any other system that allowsinteraction between the user and a remote database.

If the called party chooses to talk to the calling party over DSL, thecall processing system can use the IP address of the called party toroute the call through the network 113. One skilled in the art willrecognize that because the call is being routed over a packet switchednetwork 113 there is a bridge between the POTS circuit-switched phoneprotocol and the high-frequency packet-switched internet protocol. Thus,before passing the call through the network, the call processing systemwill translate the information into packets that will be understood bynetwork 113 and the IP phone 115.

Referring now to FIG. 2, shown is a flowchart illustrating the operationof the first embodiment, among others, of the invention. Step 200 showsthat the first step is for a T_Busy signal to be triggered. As shown instep 201, the T_Busy signal causes the system to check for an active DSLconnection at the called party computer 111. The call processing system109 then branches, as shown by step 202, according to whether or not anactive DSL connection is present. If there is no active DSL connectionpresent the call processing system 109 will either instruct theswitching system 102 to send a busy signal to the calling party orattempt to connect another service, as shown in step 203, such as voicemail.

If there is an active DSL connection present at the called partylocation, the DSL integrated call waiting service can continue toattempt to establish a connection by first retrieving information aboutthe DSL connection, such as an IP address, as shown by step 204. Firstthe call processing system 109 retrieves calling party information froma network 113 database, as shown by step 205. Then, at step 206, thecall processing system 109 sends the calling party information and/or aURL to the called party computer 111 via the DSL connection. The callprocessing system 109 then sends a prompt to the called party computer111, as shown by step 207. The called party can choose, according tostep 208, to connect the call over the DSL connection. If the partychooses not to connect the call over the DSL connection, the callprocessing system 109 will instruct the switching system 102 to send abusy signal or try another service, as shown by step 209. Alternatively,these other service choices may be moved into the prompt and decisionsteps 207, 208, respectively.

If the called party chooses to connect the call over the DSL connection,the call processing system 109 routes the call to the called party IPphone 115, as shown in step 210. This step includes translating thecircuit switched voice of POTS into a packet switched voice signal thatcan be sent over a packet network. Routing the call to the called partyIP phone 115 further includes using the DSL information retrieved fromthe network 113 in step 204 to route the packetized voice over thepacket network 113. As shown by step 211, the last step would be toconnect the call using voice over DSL after the IP phone 115 isanswered.

Referring now to FIG. 3, shown is a schematic diagram illustrating oneembodiment, among others, of the present invention. A calling partytelephone 300, such as a standard POTS telephone, can be connected tothe public switched telephone network (PSTN) 301 by line 302. The PSTN301 includes numerous service switching points (SSPs) 303, which arecentral switching offices that can connect a telephone call from thecalling party telephone 300 to a called party telephone 304 usinghandling information received from a service control point (SCP) 305.

The SCP is typically a fault tolerant computer, which preferably resideson a secured intranet 306 and contains a database which keeps track ofcall routing information and telephone customer profile information. TheSCP can be connected to an application server 307 which can beconfigured to collect information about customer applications such asvia the internet 308, for example, among others. The application server307 is preferably connected to an internet customer profile database309, which keeps track of internet customer IP and routing information.The application server 307 is further connected to a presence,preference and availability database 310 which maintains information onthe accessibility of a plurality of DSL connections. The presence isdefined according to whether or not the user's computer has an activeDSL connection registered with the application server 307. Thepreference is defined by the user and can limit access of the telephonenetwork to send information to the user's computer. Availability can bedefined as the DSL connection being registered to be present however itmay have initiated no activity for a pre-determined period of time.

The application server 307 in one embodiment, among others, can includea chat client server. The chat client server can be configured to serveas a presence and availability server 307 for a plurality of chatclients residing on subscriber computers. Thus, when a chat client isstarted on a called party computer 315, the chat client server canprovide the presence and availability information to route a callthrough the packet switched network 308 to an IP phone associated withthe computer 315.

One skilled in the art will recognize the abundance of chat clientspresently available that may be used in conjunction with the presentinvention. Some of the more popular of these chat clients include: MSNMessenger, available from Microsoft, Corp. of Redmond, Wash.; Yahoo!Messenger, available from Yahoo!, Inc. of Sunnyvale, Calif.; AOL InstantMessenger, available from America Online, Inc. of Dulles, Va.; andJabber Instant Messenger, available from Jabber, Inc. of Denver, Colo.One skilled in the art will further recognize that the Jabber InstantMessenger comprises an open systems architecture. Open systemsarchitectures generally allow a user to manipulate the source program totailor the client to specific needs of each individual user. Used inthis context, the open system architecture could facilitate thedevelopment of a custom application to be provided with the integratedchat client. One skilled in the art will further recognize the existenceof UNIX and LINUX chat programs and other programs, such as textmessaging on wireless phones, that allow text communication between twoparties. In alternative embodiments, each of these alternative textcommunication applications are intended to be included within the scopeof the present invention. Generally, as understood herein, chat clientsat least provide some type of text-based communication, in accordancewith the preferred embodiment of the invention.

The application server 307 in one embodiment, among others, of thepresent invention, can be connected to a session initiation protocol(SIP) server 311. SIP is a protocol which allows the circuit switchedtelephone call to be connected over the packet switched protocol used onmost networks. The SIP server 311 is in communication with an SIPsignaling gateway 312 and an SIP Media Gateway 313. The SIP signalinggateway 312 allows the SIP server 311 to communicate with the SS7protocol packet network included in the PSTN 301 by translating the SS7protocol to SIP protocol. The SIP Media gateway 313 can reside on theinternet 308, and is configured to receive the POTS band circuitswitched voice signal from the PSTN 301, and to convert the POTS bandcircuit switched voice signal to a packet switched signal, and convertthe packet switched voice from the computer 315 to circuit switchedvoice. The SIP server 311 sends the address information to the SIP mediagateway 313 to be used in addressing the packetized voice data signal,such that the signal will arrive at the intended subscriber.

The SIP media gateway 313 can be connected to an internet serviceprovider (ISP) 314, and is configured to send the converted packetizedvoice signal to the ISP 314. The ISP 314 provides high speed DSLinternet service to a number of subscribers over the standard PSTNtwisted pair telephone line. The ISP 314 receives the packetized voicesignal and sends it through the PSTN 301 to a DSL connected computer 315at the called party premises. Software on the DSL connected computer 315can then decode the signal and output the voice signal. Moreover, thesoftware on the DSL connected computer 315 can compress and send a voicesignal through the network to the calling party in the reverse of themanner discussed above.

The SIP server 311 works together with the SCP 305 and the applicationserver 307 to provide the called party another option for answering thephone when the called party analog phone 304 is in use. The SSP 303 hasa plurality of triggers that, when fired, will cause the SSP 303 to senda query to the SCP 305 for call handling information. When the calledparty analog phone 304 is busy, the SSP 303 triggers a T_Busy signal andit sends a query to the SCP 305. The SCP 305 is configured to check thetelephone customer database (not shown) and obtain information about howto handle the call. If the called party subscribes to DSL integratedcall waiting, the SCP 305 then queries the application server 307 tofind out if the called party has an accessible DSL connection. If not,the DSL integrated call waiting process ends and the SCP 305 may tryother services if there is no DSL connection accessible.

If the called party has an accessible DSL connection, the applicationserver 307 then retrieves the called party IP address from the customerprofiles 309 and sends a prompt to the called party computer 315. Theprompt typically could include call waiting information about thecalling party and ask the called party whether he or she would like toconnect the call over the DSL connection. The prompt could also includeoptions regarding other services to which the called party subscribes,or other information that could be personalized by the calling party.This personalization could be accomplished by using a calling partycomputer 316, to connect over the internet 308 to update a customerdefined settings database 317, which the application server 307 wouldaccess. Alternatively, the personalized information could be stored in acustomer defined settings database coupled to the application server 307on the telephone service provider intranet 306 and the calling partycould use the calling party computer 316 to adjust the database over theinternet 308. Further, one skilled in the art will recognize that aninteractive voice response (IVR) system could even facilitate theupdating of the personalized information over the telephone.

If the called party does not wish to connect the call over the DSLconnection, another service may be offered. However, if the called partywishes to connect the call over the DSL connection, the applicationserver 307 notifies the SCP 305, which instructs the SSP 303 to routethe call using the SIP server 311. The SIP server 311, however, cannotunderstand the SS7 signaling protocol, thus an SIP signaling gateway 312is used to translate the SS7 protocol into the SIP protocol. The SIPserver 311 then retrieves the IP address of the called party computer315 from the application server 307 using the signaling informationreceived from the SIP signaling gateway 312. The SIP server then passesthis information to the SIP media gateway 313. The SIP media gateway 313translates the circuit switched voice signal into a packet switchedvoice signal able to be transferred over IP, and send the packetizedvoice to the called party computer 315 using the IP address informationreceived from the application server 307.

One skilled in the art should understand that this invention is notintended to be limited to merely the SIP standard interface used in thepresent embodiment. As one skilled in the art will recognize, there areother interface protocols that could be substituted for the SIPinterface, such as media gateway control protocol and H.323. Each ofthese alternative interface protocols, among others, are intended to bewithin the scope of the present invention.

Referring now to FIG. 4, shown is an example call flow diagram accordingto one embodiment, among others, of the system of FIG. 3. To start thecall flow, in step one a call is made by a calling party to a calledparty. A connection request associated with the call arrives at thecalled party SSP 303. If the called party telephone 304 is busy, aT_busy signal will be triggered, which initiates a query to be sent fromthe called party SSP 303 to the SCP 305, according to step two.

When the SCP 305 receives the query from the SSP 303, the SCP 305 checksits database. Upon checking the telephone customer database, the SCP 305finds that the called party subscribes to a number of services forhandling calls when the line is busy, including DSL integrated callwaiting. In step three, The SCP 305 then sends a query to theapplication server 307 to check if a DSL connection associated with thecalled party telephone number is accessible. If the DSL connectionassociated with the called party telephone number is accessible, thenext step has the SCP 305 sending the calling party name and/or thecalling party number to the application server 307.

Once the application server 307 has received the calling party nameand/or number, the application server 307 can retrieve personalinformation about the calling party from its customer database, or fromanother source. After retrieving this personal information on thecalling party, the fifth step involves pushing the personal informationto the called party DSL connected computer 315 along with a prompt. Theprompt typically asks the party to choose the method of calldisposition. The choices for call disposition can typically be thoseservices to which the called party subscribes, including the DSLintegrated call waiting service. However, the telephone service providercould also provide all service choices to the called party computer 315together with prices and charge the called party based upon thedisposition of the call.

After providing the prompt, the next step has the user selecting to talkto the calling party using voice over DSL. The application server 307can then receive the called party choice and notify the SCP 305 aboutthe disposition of the call and the phone number of the calling party.In step eight, the SCP 305 instructs the SSP 303 to route the call usingthe SIP server 311. Next, the SIP server 311 can retrieve the calledparty IP phone address from the application server 307 in order to routethe call to the correct DSL connected computer 315. Finally, afterretrieving the called party IP phone address, the SIP server 311 canestablish a connection between the called party and the calling party.The called party will typically still be using a standard analog POTStelephone 300, while the called party will be using a DSL connected IPphone or soft phone, the protocol gap being bridged by the SIP signalinggateway 312 and the SIP media gateway 313.

Referring now to FIG. 5, shown is a schematic view of a secondembodiment, among others, of the present invention. A calling partyphone 500 is coupled to a central office switching station 501. Thecentral office switching station 501 is typically coupled to a pluralityof other switching stations 502, 503 by a plurality of circuit switchedlines 504, 505, 506. Each of these circuit switched lines 504, 505, 506has a plurality of 64 kbps timeslots for each of a plurality ofconnected calls. Each of the central office switching stations 501, 502,503 are further coupled to a packet signaling network 507 via packetswitched lines 508, 509, 510. The packet signaling network uses the SS7protocol, and handles the signaling and routing information for each ofthe plurality of telephone calls connected across the PSTN 511.

In order to facilitate the handling of the plurality of telephone calls,the packet signaling network 507 is coupled to a handling database 512.The handling database 512 typically keeps track of customer specificsubscription information and routing and handling information for all ofthe telephone service provider customers. Typically, the handlingdatabase 512 looks up the handling information for the called party andinstructs the central office switching station 503 to connect a callfrom the calling party phone 500 to the called party phone 513.

However, when the called party phone 513 is busy, one embodiment, amongothers, of the present invention is configured to check for a DSLconnected computer 514 at the called party premises upon receiving abusy signal at the called party telephone 513. If the DSL connectedcomputer 514 is accessible, the phone service is designed to prompt theuser to connect the call to an IP phone 515. Thus, the DSL connectionmakes available higher bandwidths to connect the call, using a DSLAM 516to multiplex high frequency data and low frequency voice together, witha modem 517 to modulate and demodulate the data signal, even though thecalled party telephone 513 is busy.

This service is provided by coupling the handling database to anapplication server, which is connected to a packet switched network,such as the internet. The application server 518 keeps track of whichnetwork subscribers have active DSL sessions on the network 519.Typically this could be done with a custom application residing on thecalled party computer 514. The custom application could sign onto theapplication server upon starting the custom application, where it can beregistered with the network customer database 520. Alternatively, theapplication server may be able to search for a called party DSLconnected computer at another site, such as an internet serviceprovider, or some database residing on the network 519 itself.

Once the application server has ascertained that the called party DSLconnected computer 514 is accessible, it can retrieve the calling partyname and telephone number from the handling database 512. Theapplication server can further retrieve other personal information aboutthe calling party, through the network 519, an internal database, orsome other database, and include the information in a prompt pushed tothe called party computer 514. Pushing the prompt can be facilitated byretrieving the called party computer 514 IP address from the networkcustomer database 520, and sending the prompt to a custom applicationresiding on the called party computer 514.

A media gateway 521 to translate the circuit switched PSTN voice to andfrom the packet switched IP is included, so as to bridge the differencesbetween the two networks. The media gateway is coupled to the centraloffice circuit switch via a circuit switched line 522. The media gatewayis also coupled to a protocol server 523, which provides routinginstructions to the media gateway for inclusion with the packets sent tothe called party computer 514. These routing instructions are receivedvia a signaling gateway 524 coupled to the protocol server 523, and areprovided to translate the signal switched SS7 protocol to the mediaprotocol. The signaling gateway 524 is coupled to the packet signalingnetwork 507 through a packet switched connection 525. These mediaprotocol are typically referred to as VoIP, and as one skilled in theart will recognize, there exist numerous varieties of such mediaprotocol that can be used in conjunction with the present invention.

Thus, when a call is placed by the calling party telephone 500, it isreceived at the central office circuit switch 501. The central officethen sends a packet switched signal to the signaling network 507,indicating a call request has been received and requesting handlinginstructions. The signaling network 507 then relays this information tothe handling database 512. The handling database notifies central officecircuit switch 503 of the incoming call, and receives a query from thecircuit switch 503 indicating that the called party telephone is busy.The handling database then examines the telephone customer profile andfinds that the called party subscribes to DSL integrated call waiting.The database then queries the application server 518 to obtain theaccessibility of a DSL connected called party computer 514.

If the DSL connected called party computer 514 is accessible, theapplication server sends a prompt to the called party computer 514. Theprompt typically includes a variety of calling party informationobtained through a coupled database. When the called party chooses totalk over the DSL connection, the application server 518 notifies thehandling database 512 of the disposition, which in turn instructs thecentral office circuit switch 503 of the called party to route the callvia the DSL line. The central office circuit switch 503 then routes thecall over line 522 to the media gateway 521. The packet signalingnetwork 507 routes the handling information over line 525 to thesignaling gateway 524. The signaling gateway 524 translates the SS7packets to the correct media and send the translated packets to theprotocol server 523. The protocol server 523 in turn instructs the mediagateway where to send the translated voice packets. The translated voicepackets are then sent to the DSL connected called party computer 514 viathe DSLAM 516 and the modem 517. The computer 514 or IP phone 515 thendecodes and plays the packetized voice signal. Similarly, the computer514 or IP phone 515 can also receive a called party voice. This calledparty voice is then encoded to be sent back through the network 519,translated, and transported by the PSTN to the calling party telephone500.

The systems of the preferred embodiments described herein would allow acalled party to answer a telephone call over a DSL connection. Oneadvantage such systems could provide is the ability to provide a secondtelephone connection for incoming calls. A further advantage would bethat the called party can maintain a conversation on the analogtelephone line while connecting a telephone call on an IP phone. Forexample such a service could be invaluable to a parent who wishes toreceive telephone calls while a son or daughter continues a conversationon the standard phone line, while only limiting outgoing calls.Moreover, this is done without necessitating the initiation of a secondstandard telephone line. Furthermore, the DSL integrated call waitingservice would enable the called party to continue to use the internetfunctionality of the DSL connection. Thus, the called party can maintaintwo telephone conversations while experiencing only a relatively smallamount of bandwidth loss over the DSL connection due to the displacementof the second voice call.

Process and function descriptions and blocks in flow charts can beunderstood as representing, in some embodiments, modules, segments, orportions of code which include one or more executable instructions forimplementing specific logical functions or steps in the process, andalternate implementations are included within the scope of the preferredembodiment of the present invention in which functions may be executedout of order from that shown or discussed, including substantiallyconcurrently or in reverse order, depending on the functionalityinvolved, as would be understood by those reasonably skilled in the artof the present invention. In addition, such functional elements can beimplemented as logic embodied in hardware, software, firmware, or acombination thereof, among others. In some embodiments involvingsoftware implementations, such software comprises an ordered listing ofexecutable instructions for implementing logical functions and can beembodied in any computer-readable medium for use by or in connectionwith an instruction execution system, apparatus, or device, such as acomputer-based system, processor-containing system, or other system thatcan fetch the instructions from the instruction execution system,apparatus, or device and execute the instructions. In the context ofthis document, a computer-readable medium can be any means that cancontain, store, communicate, propagate, or transport the software foruse by or in connection with the instruction execution system,apparatus, or device.

It should be emphasized that the above-described embodiments of thepresent invention are merely possible examples of implementations setforth for a clear understanding of the principles of the invention. Manyvariations and modifications may be made to the above-describedembodiment(s) of the invention without departing substantially from theprinciples of the invention. All such modifications and variations areintended to be included herein within the scope of this disclosure andthe present invention and protected by the following claims.

1. A call handling system for tracking customer specific subscription information and routing and handling information for telephone service provider customers, the call handling system comprising: a handling database system configured to instruct a switching system to connect an analog call from a calling party to the called party; receive a query from the switching system indicating that a called party telephone is busy; in response to the query, examine handling information associated with the called party to determine if the called party has voice over Internet Protocol (VoIP) service; and instruct the switching system to route the analog call over an Internet connection to the called party after the called party chooses to communicate using the VoIP service.
 2. The call handling system of claim 1, further comprising: an application server configured to receive a query from the handling database system to obtain an accessibility of the VoIP service of the called party.
 3. The call handling system of claim 2, wherein the application server is further configured to send a prompt to a called party computer asking the called party to choose to receive an incoming analog call using the VoIP service.
 4. The call handling system of claim 3, further comprising: a media gateway residing on the Internet and configured to receive the analog call and convert the analog call received from a PSTN network to a digital call transmitted via the Internet.
 5. The call handling system of claim 4, wherein the analog call follows a SS7 protocol and the digital call follows a SIP protocol.
 6. The call handling system of claim 4, wherein the analog call is routed to the called party using VoIP service while retaining a prior analog call to the called party over a PSTN network, the prior analog call causing the busy indication.
 7. A call handling method comprising: receiving a query from a switching system indicating that a called party telephone is busy; examining handling information for the called party to determine if the called party has voice over Internet Protocol (VoIP) service; and instructing the switching system to route an analog call over a VoIP connection to the called party after the called party chooses to communicate using the VoIP service.
 8. The call handling method of claim 7, further comprising: requesting accessibility status of the VoIP service of the called party.
 9. The call handling method of claim 8, further comprising: sending a prompt to a called party computer asking the called party to choose to receive an incoming analog call using the VoIP service.
 10. The call handling method of claim 9, further comprising: receiving the analog call and converting the analog call received from a PSTN network to a digital call transmitted via the Internet.
 11. The call handling method of claim 10, wherein the analog call follows a SS7 protocol and the digital call follows a SIP protocol.
 12. The call handling method of claim 10, wherein the analog call is routed to the called party using VoIP service while retaining a prior analog call to the called party over a PSTN network, the prior analog call causing the busy indication.
 13. The call handling method of claim 10, wherein the accessibility status is based upon preference information stored in a database, the preference information indicating whether the called party prefers to receive voice calls using VoIP.
 14. A computer readable medium having a program for handling a telephone call, the program comprising: receiving a query from a switching system indicating that a called party telephone is busy; examining handling information for the called party to determine if the called party has voice over Internet Protocol (VoIP) service; and instructing the switching system to route an analog call over a VoIP connection to the called party after the called party chooses to communicate using the VoIP service.
 15. The computer readable medium of claim 14, the program further comprising: requesting accessibility status of the VoIP service of the called party.
 16. The computer readable medium of claim 15, the program further comprising: sending a prompt to a called party computer asking the called party to choose to receive an incoming analog call using the VoIP service.
 17. The computer readable medium of claim 16, the program further comprising: receiving the analog call and converting the analog call received from a PSTN network to a digital call transmitted via the Internet.
 18. The computer readable medium of claim 17, wherein the analog call follows a SS7 protocol and the digital call follows a SIP protocol.
 19. The computer readable medium of claim 17, wherein the analog call is routed to the called party using VoIP service while retaining a prior analog call to the called party over a PSTN network, the prior analog call causing the busy indication.
 20. The computer readable medium of claim 17, wherein the accessibility status is based upon preference information stored in a database, the preference information indicating whether the called party prefers to receive voice calls using VoIP. 